Methods for transporting digital media

ABSTRACT

A networked system is provided for transporting digital media packets, such as audio and video. The network includes network devices interconnected to send and receive packets. Each network device can receive and transmit media signals from media devices. A master clock generates a system time signal that the network devices use, together with a network time protocol to generate a local clock signal synchronised to the system time signal for both rate and offset. The local clock signal governs both the rate and offset of the received or transmitted media signals. The system, which can be implemented using conventional network equipment enables media signals to be transported to meet quality and timing requirements for high quality audio and video reproduction.

CROSS REFERENCE TO RELATED APPLICATION

This application is a continuation of U.S. patent application Ser. No.16/580,725, filed Sep. 24, 2019, which is a continuation of U.S. patentapplication Ser. No. 16/119,595, filed Aug. 31, 2018, now U.S. Pat. No.10,461,872, which is a continuation of U.S. patent application Ser. No.15/187,309, filed Jun. 20, 2016, now U.S. Pat. No. 10,097,296, which isa continuation of U.S. patent application Ser. No. 14/636,953, filedMar. 3, 2015, now U.S. Pat. No. 9,398,091, which is a continuation ofU.S. patent application Ser. No. 13/907,171, filed May 31, 2013, nowU.S. Pat. No. 9,003,009, which is a continuation of U.S. patentapplication Ser. No. 13/193,283, filed Jul. 28, 2011, now U.S. Pat. No.8,478,856, which is a continuation application of U.S. patentapplication Ser. No. 12/662,994, filed May 14, 2010, now U.S. Pat. No.8,005,939, which is a continuation application of U.S. patentapplication Ser. No. 11/409,190, filed Apr. 26, 2006, now U.S. Pat. No.7,747,725, which claims priority from Australian Provisional ApplicationNo. 2005902065, filed Apr. 22, 2005, and Australian ProvisionalApplication No. 2005906272, filed Nov. 11, 2005. Each of theseapplications, in their entirety, are incorporated herein by reference.

BACKGROUND OF THE INVENTION Technical Field

This invention concerns the transporting of digital media packets, suchas audio and video. In a first aspect the invention concerns a datanetwork able to carry high fidelity audio. In another aspect theinvention concerns a network device for sending and receiving digitalmedia packets; such as a Digital Signal Processing (DSP) chip for publicaddress systems. In a further aspect the invention concerns a method ofoperating the data network or network devices and software forperforming the method, for instance, on a personal computer.

Background Art

Audio and video signals have long been transmitted using applicationspecific cables. For instance two-core speaker cable is used to carryleft and right audio signals from amplifiers to speakers.

The time at which a received media signals are played out by a mediadevice is called the playout time. Typically, a media device thatreceives media signals will playout the media signals by rendering themin some way. For example, if the media device is a loudspeaker it willrender the audio media signal into sound. If the media device is a videoscreen it will render frames of the video media signal onto a screen.Alternatively, if the media device is a lighting control system it willrender the lighting media signal by turning a spotlight on and off.

Real time transporting of digital audio and video and other digitalmedia over data networks creates a new set of problems compared tonon-media data. For instance, data networks may use packet switching inwhich data is divided into packets for separate transmission. As thepackets are transmitted, sequential packets may take different routesand have different transit times. The packets are numbered to ensurethey can be reordered correctly after arrival. This technique, however,does not suffice when left and right audio signals are to be received atdifferent destinations, for instance at different speakers.

Unlike non-media data, digital media must be played out insynchronisation. For example, video and audio must be aligned in time sothat when they are played out the images match the sound.

The concept of a network clock has been used to address timing problemsin data networks. A network clock signal is typically generated at aspecific point in the network and this becomes the system time signalreceived by devices on the network. The system time signal is then usedas a time reference for every device that receives the system timesignal. Because of the topology of the network, devices at differentlocations on the network will receive the clock signal with a phaseoffset from the network clock, depending on the propagation delay fromthe clock to the device. A further consequence is that the differentremote devices will have received clock signals that have phase offsetswith respect to each other, as well as with respect to the networkclock.

Digital media transmission has historically embedded clockinginformation in the transmitted data. Embedding and recovering clockinginformation from data signal transitions or packet timing (e.g. AES3,SP/DIF, Gibson MaGIC) works well for point to point links between asmall numbers of devices, but as the number of devices increases, clockjitter cascades and builds through each device that recovers andre-transmits the clocking information. Large systems employ a separateclock network to avoid such problems with clock jitter.

Digital media transmissions may alternatively employ a Time DivisionMultiplexing (TDM) approach. In TDM systems (e.g. MADI, CobraNet), amaster clock device initiates periodic transmission cycles and eachdevice is allocated one or more time slots within that cycle fortransmission. This limits the total available number of channels.

BRIEF SUMMARY OF THE INVENTION

The invention provides a method of transporting multiple media channelsover a network in a manner that meets quality and timing requirementsfor high quality audio and video reproduction without the need forspecialised equipment, such as a separate clock network. The inventionhas the advantage of being able to use low cost data network cabling andequipment, such as switched Ethernet, to transport digital media.

The invention provides network devices having a local clock signal thatis synchronised to a system time signal of the network for both rate andoffset. Rate synchronisation helps to ensure the rate that the localclock ticks over at is the same as the rate of the system time signal.Offset synchronisation helps to ensure both the system time signal andlocal time signal share a common time reference point. The local clocksignal governs the rate and offset of the received or transmitted mediasignals. Because each network device has a local clock signal that issynchronised to the system time signal, all received and transmittedmedia signals of the network are synchronised for both rate and offset.

Accordingly, it is a feature of this invention to provide a network andsuitable devices for transporting digital media. The network includes amaster clock device to generate a system time signal for the network anda plural number of network devices interconnected such that each deviceis able to send packets to other network devices, and to receive packetsfrom other network devices. Preferably, each network device is alsoconfigured to receive media signals from a media device, to transmitmedia signals to a media device, or to do both. Typically, each networkdevice is connected to the master clock device and uses the system timesignal and a network time protocol to generate a local clock signal thatis synchronised to the system time signal for both rate and offset, thelocal clock signal governing both the rate and offset of the received ortransmitted media signals. The media packets may be audio or video, or amixture of both.

Typically, each network device that receives media signals from a mediadevice uses its local clock which is synchronised to the system timesignal to packetise the media signals, and to timestamp the mediapackets with system time before sending them to another network device.In this way, media packets are generated and sent by all network devicesin the network at a local rate that is synchronised to the rate of thesystem time signal. This helps to eliminate buffer overrun and underrunin the network. Further, the packets are timestamped with a synchronisedtime value. The timestamp may be the earliest system time at which themedia signal contained in the packet was converted into digital form(i.e. generation time).

Each network device that receives media packets uses the packettimestamp to coordinate the media signals in time for playout. Thecoordination of media signals is achieved using the packet timestamps tore-order the media packets if required; and in addition to align andcombine media signals received from different network devices, or todetermine the appropriate playout time for media signals contained inthe media packets, or both. Aligning media signals may comprise delayingthe playout of one media signal with respect to another.

The appropriate playout time for media signals contained in a mediapacket may be different for each network device that receives the mediapacket. In this way, where there are multiple playout media devices, theplayout time can be controlled based on the spatial location of eachplayout media device, so that media signals are played out by each mediadevice is received at one particular spatial location at the same time.For, example in a rock concert the sound from the speakers on stage andin the auditorium should arrive at the audience at the same time. Toachieve this, sound from the closer auditorium speakers can be delayedto allow the sound from the stage speakers to travel the extra distancethrough the air.

A media clock signal may be provided that is synthesized from the localclock signal. The media clock signal is able to drive conversion betweenanalogue and digital media signals and to directly govern the rate andoffset at which media signals are received or transmitted (e.g. areproduced or consumed) by the network device. Because the media clock issynthesised from the local clock signal, which is in turn synchronisedto the system time signal, the rate that media signals are produced andconsumed is governed in a manner that eliminates buffer overrun andunderrun in the network devices.

Techniques that can be applied to synthesise a media clock from thelocal clock signal include Direct Digital Synthesis (DDS), DigitallyControlled Oscillators (DCO) or Voltage Controlled Oscillators (VCO)controlled by a Digital to Analogue Converter (DAC). The media clock maybe synthesized by using a software timer that is caused to track thesynchronised local clock, to directly govern the rate and offset of thereceived or transmitted media signals.

The rate of the media clock signal may be different to the rate of thelocal clock. For example, the rate of the media clock may be a multipleof the rate at which the media signals are generated by the mediadevice, for example a multiple of the rate at which digital audiosamples are produced. Use of a network time protocol to synchronise thelocal time to the system time signal on the network de-couples clockingand clock synchronization from data transmission. This simplifiessupport for simultaneously transmitting at multiple data rates, such as44.1 kHz/48 kHz/96 kHz audio by different media clocks, and eliminatesthe need to run a separate clock network to each network device. Becausethe media clock can derive any sample rate or frame rate it likes fromthe system time signal, the network as a whole is not constrained to asingle sample rate. One or more media devices may be connected to anetwork device to generate and deliver media signals, to receive, toprocess and to deliver media signals, or to receive and playout mediasignals.

The jitter experienced by network time protocol packets is minimised byusing a Quality of Service (QoS) technique. The QoS technique mayoperate in the network to expedite the forwarding of packets having ahigher priority than other packets. For example, packets may beprioritised based on three categories. Packets that contain network timeinformation may be prioritised over audio media packets. In turn, audiomedia packets may be prioritised over best effort traffic. By placingthe highest priority on packets that contain network time information,the QoS scheme is able to optimise the maintenance of the timingsynchronising of the network devices.

According to one embodiment of this invention, if two network devicesseek to send media packets to another network device, they each mayselect a random start time and send media packets from their respectiverandom start time at a regular rate. In this way, two network devicessending media signals to be played in synchronization do not cause abottleneck in the network that would otherwise increase the jitter inthe network.

The network time protocol (e.g. according to IEEE 1588) distributesabsolute time from a master clock. The network time protocol may use abi-directional exchange of messages to enable the calculation of thetransmission delay between master and local slave clocks which can beused to calculate a more accurate estimate of the local clock offset.This enables the local clocks to compensate for variable network delayand achieve tighter synchronization regardless of their location in thenetwork topology.

The network time protocol may operate to elect a master clock devicewhere multiple possible master clock devices exist on the network. Forexample, in the IEEE 1588 network protocol, the best master clock andbackup master clock are selected from a set of potential candidates. Thebackup master clock uses the IEEE 1588 network time protocol tosynchronise the backup clock with the elected master clock. If theselected master clock fails, the backup master clock operates to takeover by providing the system time signal for the network. Networkdevices that depend on a master clock can continue to operate smoothly.

Further, the network time protocol may support multiple master clockdevices such that different network devices are connected to differentmaster clocks. For example, the data network may be comprised of twodistinct audio networks in different locations each having a IEEE 1588master clock. The distinct networks may coordinate their master clocksvia another mechanism, for example GPS time synchronisation. This allowsthe two networks to share a common notion of time and send timestampedmedia packets to each other.

The network may include an Ethernet network. Ethernet networks are ableto support multicast transmission in which network packets areduplicated in network switches, thus efficiently supporting largenumbers of receiving network devices. The network devices may beconnected directly, or indirectly to the master clock. The architectureof this invention scales with increasing Ethernet bandwidth (e.g. 100Mbit/second to 1 Gbit/second). Increasing bandwidth implies increasedtotal throughput as well as decreased latency and jitter. Aheterogeneous network combining network devices connected via 100Mbit/sec links interconnected by 1 Gbit/sec trunks is also enabled.

The playout time for media signals must take account of networktransmission delays, network time protocol synchronization errors, mediaclock synthesis errors, sender timer jitter and network jitter. Theseare all factors which may delay or produce the appearance of the delayin the receiving of media packets. The playout time of received mediasignals must be delayed enough to allow for late arrival of mediapackets due to any of these causes since if a playout time is selectedthat is too early, any delayed media packets containing media signalsthat must be played out in synchronization with the received mediasignals will not be available for playout when required. The mediapackets are buffered until their playout time in order to compensate fordelays in transmission and clock synchronization errors. Each networkdevice may include a jitter compensation buffer provisioned according tothe maximum expected end-to-end delay such that there is never a bufferunderrun. Media packets arriving are stored in the jitter compensationdata buffer until the correct playout time arrives.

In another aspect the invention provides a data network device forconnection to other network devices to transport media packets betweenthem. The network device includes a media port to receive media signalsfrom a media device, to transmit media signals to a media device, orboth; a master clock to generate a system time signal, or a clock portto receive a system time signal; and a local clock generator to use thesystem time signal and a network time protocol to generate a local clocksignal that is synchronised to the system time signal for both rate andoffset, the local clock signal governing both the rate and offset of thereceived or transmitted media signals.

In use, when the network device receives media signals from a mediadevice it may use its local clock to packetise the media signals, and totimestamp the media packets with system time before transmitting them toanother network device.

In a further aspect, the invention provides a method of operating a datanetwork device to transport media packets. The method includes steps ofgenerating or receiving a system time signal for the network, andconfiguring a plural number of the network devices such that each isable to send packets to and/or receive packets from other networkdevices. It further includes steps of configuring the network devices toreceive media signals from a media device, transmitting media signals toa media device, or both, and connecting each network device to receivethe system time signal. Finally a step of operating each network deviceto use the system time signal and a NTP to generate a local clock signalthat is synchronised to the system time signal for both rate and offset,the local clock signal governing the rate and offset of the received ortransmitted media signals is performed.

The method may further comprise receiving media signals at a networkdevice, packetising the media signals and timestamping the media packetswith system time, using the device's local clock, and then transmittingthe media packets over the network to another network device. The methodmay further comprise receiving a media packet at a network device andusing the packet timestamp to reorder media packets; and, to align andcombine media signals from different network devices, or to determinethe appropriate playout time for media signals contained in the mediapackets, or both. If desired a software program can be used to implementthe method as described above.

BRIEF DESCRIPTION OF THE DRAWINGS

Examples of the invention will now be described with reference to theaccompanying drawings, in which:

FIG. 1 is a schematic diagram of a simple network using the invention;

FIG. 2 is a block diagram of a network device that is able to receive,send and process media packets;

FIG. 3 is a block diagram of a programmable logic device configured fordirect clock synthesis;

FIG. 4 is a clock diagram showing PMW modulation clock synthesistechnique;

FIG. 5 is a waveform diagram, showing PWM modulation;

FIG. 6 is a waveform diagram showing the effect of the local highfrequency clock being interrupted; and

FIG. 7 is a block diagram of a typical implementation of the invention.

DETAILED DESCRIPTION OF THE INVENTION

Overview of the Components of the Network

Referring first to FIGS. 1 and 2 , a data network 100 comprises a masterclock device 102 to generate a system time signal 104 for the network100. Two network devices 108 and 110 are connected to each other by anetwork 106 so that they are able to send and receive media packets. Thenetwork devices 108 and 110 are also connected to media devices 112 and114 respectively which are able to generate and/or playout mediasignals. The network device 108 is able to receive and packetize mediasignals to be sent as media packets over the network 106. The networkdevice 110 is able to receive pacts and transmit media signals containedin the media packets to the media device 114. Network devices 108 and110 each contain a local clock 122 and 124 and a media clock 123 and 125respectively.

The network includes a network time protocol (NTP) 120. A NTP 120 is aset of network messages used to synchronise a clock of one device with aclock of another device. In this case, the local clocks 122 and 124 aresynchronised with the master clock 102 using the NTP 120 and the systemtime signal 104. The network messages sent by the NTP 120 includes thesending of packets on the network 106 that relate to the system time.There are various known standard NTPs, for example, the IEEE 1588Precision Time Protocol, and the IETF NTP.

Media clock signals 130 and 132 are derived (i.e. synthesized) from thelocal clock signals (i.e. Local TOD signal) 126 and 128 respectively.The NTP uses a bi-directional exchange of messages to enable thecalculation of both clock offset and rate.

Clock rate synchronisation ensures that the rate at which the networkdevices 108 and 110 send and/or receive data packets is the same towithin a desired accuracy. Clock offset synchronisation ensures that thetime difference from the master clock 102 to the local clocks 122 and124 is the same to within a desired accuracy. In this way any two clocksin the network have a bounded rate and offset error. Ratesynchronisation also ensures that the media signal is produced andconsumed by the network devices at a rate that is derived from the rateof the master clock. The derived rate of the network device may bedifferent from the rate derived by another network device on thenetwork. The derived rate (local clock frequency) may be related to thesample rate of the media signals that the network device is convertinginto packets (i.e. 256 for 48 Hz or 256 for 44.1 Hz). Rate and offsetsynchronisation enables the local clocks 122 and 124 to compensate forvariable delays (i.e. days in the reception of media packets) andachieve tighter synchronisation regardless of the location of thenetwork device 108 and 110 in the network 100 topology.

The synchronized local clock 122 is used to timestamp media packets withthe earliest system time (that is also the local time 126) at which themedia signal contained in the packet was converted into digital form bythe data converter 140. Using the timestamp, the network device 110 thatreceives the media packet can coordinate the correct playout time forthe media signals.

Overview of the Operation of the Network

In reference to FIG. 1 , operation of the sending network device 108will now be described. Incoming analogue media signals produced by themedia player 112 reach the analogue to digital converter 140 of thenetwork device 108. The rate that the analogue to digital converter 140converts the media signal is governed by the media clock 130. Thedigital signal produced by the analogue to digital converter 140 ispassed to a data packetise and timestamp buffer 142 for collection intomedia packets. The media packets are timestamped with the earliestsystem time at which the media signal contained in the packet wasconverted into digital form.

The local clock 122 supplies rate control, and offset control to thepacketising and time stamping of the media packets via link 130. Sincethe local clock 122 is synchronised with the master clock 102, the ratethat the media packets are produced is at the same rate as the systemtime signal 104 of the master clock 102. This will also be the same rateof the local clock 124 of the receiving network device 110 so the ratethat the entire network 100 produces and consumes media packets issynchronised. The local clock 122 is also synchronised to the masterclock 102 for offset. The adjustment of the local time to synchronisewith the time of the master clock 102 is achieved using an offsetamount. The local time offset from some epoch (e.g. seconds since00:00:00 Jan. 1, 1970) tracks the master clock time. The packets arethen passed to the network port 144 for transmission on the network 106for delivery to the network device 110.

Operation of the receiving network device 110 will now be described.Incoming packets are received from the network 106 in a jittercompensation data buffer 148 (described in more detail below) where theyare delayed to account for the maximum expected latency variation (orjitter) between the sender 108 and the receiver 110. The receiver 110uses the time stamps of the received packets to reorder the packets ifnecessary. The receiver 110 may align and combine media signals receivedfrom different sources. Further, the device 110 also determines theplayout time for the media signals. The media signals are then passed tothe digital to analogue converter 150 for conversion to analogue at arate controlled by the media clock signal 132 of the media clock 125.The media signal is then sent to a media device 114, for instance, forplayout.

The device 111 shown in FIG. 2 is able to perform the functions of bothnetwork devices 108 and 110. Further, this network device 111 can beused for processing media signals in a digital form. In this case mediapackets are received from one or more senders at the network port 144and processed within the network device 111 at processor 146. Thetimestamps of the received media packets are used to align the digitalmedia signals of the packets in time, if necessary. Processing takesplace to produce a new set of digital media signals (e.g. mixing a setof audio channels to a left/right stereo channel pair). This processingtakes place at the rate and offset controlled by the local clock 122/124via link 149. The new digital media signals are placed into packets andtransmitted at a rate determined by the local clock signal 149 and withtimestamps that are determined by the local clock offset. The processedpackets are then retransmitted from the network port 144. Rather thangenerating a new timestamp for outgoing packets from the local clock 122and 124, it is also possible to copy a timestamp (i.e. the offset) froman incoming packet to the outgoing packet, thus preserving the time thatthe media signal was originally generated. In this way, media signalscan be processed purely in a computer domain by the components enclosedby 147 which may be part of a personal computer.

Any clock left to itself will tick along at a certain rate which will beslightly different for each piece of hardware. The process ofdisciplining a clock adjusts the rate and offset of the clock to trackanother reference clock (in this case the master clock 102). The processof clock synchronisation and synthesis will now be described in moredetail.

Local Clock Rate Synchronization

The local clock signal 126 is a local representation of time at thenetwork device 108. The local clock 122 and 124 is synchronised to thedata network master clock 102 using the NTP 120. The local clock signals126 and 128 are generated by a local oscillator. Even if each of thelocal oscillators have the same nominal frequency (e.g. 12.288 Mhz),their actual frequency (or rate) may be slightly different. In additionthis rate may drift with time due to effects such as ambient temperaturevariations. A local clock 122 and 124 is considered synchronized to themaster clock 102 if its rate is actually the same as the master clock102 to within a desired accuracy.

The local clock signal 126 and 128 may be an electrical signal (e.g.produced by a Voltage Control Oscillator (VCO)) or it may be representedas software counters maintained by an operating system date/timefacility. In all cases, the local clock signal 126 and 128 is governed(i.e. disciplined) so that it advances (ticks) at the same rate as themaster clock 102.

Local Clock Offset Synchronization

The local clock signal 126 drives a time of day (TOD) clock which can beexpressed as a number of seconds since an epoch. For example, NetworkTime Protocol version 3 (NTPv3), expresses time as two 32 bit numberscorresponding to the number of seconds and fraction of a second elapsedsince 00:00:00 Jan. 1, 1900. The TOD clock is used to timestamp mediapackets. The epoch for this TOD clock is a global constant for thenetwork. The availability of a global timestamp enables media signalsoriginating from different sources to be time-aligned and combinedaccurately by the receiving network device 110, for example, forplayout. A local clock 122 and 124 is considered to be offsetsynchronized to the master clock 102 if its time difference from themaster clock 102 is the same to within a desired accuracy. If the timedifference between master clock 102 and any local clock 122 and 124 isless than a ¼ of the sample period (5 μs for a 48 Khz signal) then anytwo local clocks 122 and 124 are synchronised to each other to within ½of the sample period. This enables samples generated simultaneously attwo different sources 112 to be accurately aligned for playout.

When the local clock signal 126 is an electrical signal (e.g. from aVCO), clock pulses increment a counter value which represents absolutetime. The counter can be read to produce a timestamp which can becompared with timestamps from the master clock 102 (via the network timeprotocol 120) in a phase locked loop to achieve absolute time (offset)synchronisation in addition to rate synchronisation.

Accurate clock offset synchronization requires a network timesynchronization protocol with a two way exchange of messages. Thisenables the local clock 122 and 124 to calculate the network delaybetween it and the master clock 102 and compensate for it.

Two standard network time protocols 120 which can be used with thisinvention are the Network Time Protocol (NTP), and the IEEE 1588Precision Time Protocol, NTP Version 3 is widely implemented and hasbeen documented by the Internet Engineering Task Force (IETF) in RFC1305as an internet draft standard. Additional NTP information can be foundat http://www.ntp.org/. IEEE 1588 is a published standard of the IEEE(Std 1588-2002) and is available from http://standards.ieee.org/.Additional IEEE 1588 information can be found athttp://ieee1588.nist.gov/.

IETF NTP 120 is used on general purpose computer systems and can readilyachieve millisecond time synchronization accuracy in a local areanetwork. IETF NTP 120 is slave driven in the sense that a request fromthe slave (i.e. here the network device 108 or 110) results in a two waymessage exchange, which enables the slave to calculate both its timeoffset from the master clock 102 and the network delay.

Herein, IEEE 1588 is the preferred time synchronisation protocol 120,although many NTPs may also be used. In general, timestamping packets asclose to the network transmission or reception time as possible reducesthe error due to end-system jitter. Timestamping can be performed (inpreference order) in hardware, in a device driver or in an application.

IEEE 1588 was designed for use with industrial control and measurementsystems and is suited to accurate hardware implementation. Hardwareimplementations have been shown to achieve sub-microsecond timesynchronization accuracy. Implementation of IEEE 1588 Border Clocks inswitches eliminates the jittering of clock packets as they pass throughthem. IEEE 1588 uses frequent multicast messages from the master tocalculate the offset. It uses a less frequent delay request message fromthe slave, resulting in a delay response message from the master tocalculate the delay.

It may also support redundant master clocks by including a mechanismwhere another master clock takes over if the original master clock 102fails. The invention can use the combination of two separate IEEE 1588networks each having their own master clock, for example, a recordingstudio in Sydney, Australia and a recording studio in Melbourne,Australia. These two networks may coordinate their master clocks viaanother mechanism, for example GPS time synchronisation. This allows thetwo networks to share a common notion of time and send timestampedpackets between each other.

The system may also be implemented using a set of network devices whichsynchronize with one master, while another set of devices synchronizewith a different master on the same network, either at different timesor simultaneously. As an example, a set-top box synchronized to anexternal video source may act as a master clock when viewing a video,whereas the amplifier in an audio system may act as the master clock formusic.

The network time protocol 120 exchanges messages at a regular rate (e.g.every 1 second). By making this interval a non-multiple of media samplerates (i.e. 48 khz or 44.1 Khz) the possibility of the clock packetsbeing encountered and being jittered by a media packet in transmissionon the network 106 is minimised.

The NTP IEEE 1588 is discussed further next. In the IEEE 1588 protocol,the best master clock 102 and a backup master clock (not shown) areelected from a set of potential candidates. The elected master clock 102may be a local clock 122 of a network device 108. The backup masterclock uses the IEEE 1588 network time protocol 129 to synchronise itsclock with the elected master 102. In the case that the master fails102, the backup master takes over and other local clocks that werepreviously synchronised with the elected master clock 102 will nowsynchronise with the backup master clock and the network 100 continuesto operate smoothly.

Media Clock Synthesis

Clock synthesis is a widely studied problem and various techniques canbe applied to synthesise a media clock 123 and 125 from the local clocksignal, including Direct Digital Synthesis (DDS), Digitally ControlledOscillators (DCO) or Voltage Controlled Oscillators (VCO) controlled bya Digital to Analogue Converter (DAC).

A digital media signal has an implied clock. For an audio media signal,this can be the audio sample rate (e.g. 48 kHz). For a video mediasignal, this can be the number of frames per second. Hardware thatproduces or consumes digital media signals often uses a multiple of thesample rate or frame rate. For example, analogue-to-digital converters140 commonly need a clock that is 128× or 256× the rate at which audiosamples are actually produced. We term this clock multiple the mediaclock 123 and 125. Media clock synthesis is the process of deriving amedia clock signals 130 and 132, that is a digital to analog converter(DAC) word clock, from the network time protocol 120.

At least three techniques can be used to derive a media clock: (i)directly disciplining a hardware clock from the network time protocol,(ii) controlling a pulse width modulated media clock 123 and 125 with amaster clock 102 that has been disciplined by NTP/1588, and (iii)controlling a software timer from a master clock 102. These techniquesare discussed next.

(i) Direct Clock Synthesis

Direct media clock synthesis can be implemented by using a programmablelogic device such as an field programmable gate array (FPGA). Referringto FIG. 3 , the frequency synthesizer block 200 generates the localclock f_(LOCAL) from a clock source f_(SRC) 199 under the control of thenetwork time protocol 120. The ratio of the nominal frequencies off_(SRC)/f_(LOCAL) is preferably greater than two. Preferably, either oftwo possible implementations of the frequency synthesis block 200 togenerate the local clock frequency f_(LOCAL) can be used. Bothimplementations use an M bit accumulator.

In a first implementation, the most significant bit (MSB) of the M bitaccumulator is a square wave of frequency f_(LOCAL). This waveform isjittered (i.e. delayed), however, by 1/f_(SRC) whenever the accumulatorbuilds up enough phase error to output an extra pulse. If f_(SRC) isselected to be an integer multiple of f_(LOCAL) then these phase shiftsare required only to compensate for the frequency offset from nominaland drift of the source clock f_(SRC) 199. These occur at a very lowfrequency and while they are large, have been shown to be inaudiblejitter.

In the second implementation very low jitter f_(LOCAL) can besynthesized by using “Direct Digital Synthesis” (DDS). DDS involvesusing P significant bits of the M bit phase accumulator to address asine wave lookup table. This is then input to a DAC 208 whose output isan analog sine wave of frequency f_(LOCAL). This sine wave is filteredto remove harmonics and a comparator is used to generate a local clockwith very low jitter. The frequency of f_(LOCAL) can be tuned to theresolution of 2^(M).

For both implementations, the output of the frequency synthesis block200 is the local clock f_(LOCAL). This signal is used to clock an N bitcounter 202. The N bit counter 202 provides the local TOD clock and isused by the timestamper 204 to timestamp sent and received network timeprotocol and media packets. The local clock frequency is further dividedby the frequency divider 206 to generate the media clock f_(MEDIA),which drives the ADC/DAC 208. Alternatively, the local clock signalf_(LOCAL) could be provided directly to the ADC/DAC 208 from thefrequency synthesis block. In this case there would be no separate mediaclock signal. Instead, the local clock signal could be used to controlthe rate that media signals are produced and consumed by the networkdevice.

The method is as follows:

-   -   1. The network time protocol runs at T second intervals and        calculates the offset O between the local and master clocks.        This offset may be a result of both a time offset between the        local and master clock, and an offset error due to a frequency        difference between the two clocks. Calculate a frequency        f_(LOCAL) at which the local clock needs to run at over the next        T seconds to compensate for this error as:        f _(LOCAL) =f _(LOCAL_NOMINAL) +O/T

2. Calculate a phase accumulator tuning wordTW=2^(M) *f _(LOCAL) /f _(SRC)

-   -   where TW is the input to the M bit accumulator 200 and is added        to its output every 1/f_(SRC) seconds. The NTP protocol time        processes the network time protocol messages and uses the        locally generated transmit and receive timestamps to compute an        updated tuning word (TW). The tuning word adjustments may be        smoothed to remove the impact of jitter in the time offsets due        to network jitter by using a PLL, instead of making        instantaneous adjustments.

(ii) Pulse Width Modulation (PWM) Clock Synthesis

The PWM clock synthesis technique for media clock synthesis can usecommercially available digital processor (DSP) chips 220 and 222 (seeFIG. 4 ), a time of day clock disciplined by NTP or IEEE 1588 as foundin many embedded operating systems, a local high frequency (f_(SRC) Hz)clock source 199, a PWM counter output pin and a programmable counterthat can generate interrupts every P 180 cycles of the local highfrequency clock. The PWM counter can be programmed to repeatedly outputlogic HI for a count of M cycles of f_(SRC), then logic LOW for N cyclesof f_(SRC).

As shown in FIGS. 5 and 6 , the method involves:

1. Estimate, as described above, the frequency of the local highfrequency clock f_(SRC) 199 using the programmable counter 220. Forexample, if the programmable counter generates an interrupt every Pclock cycles and the system time of day clock is recorded each time aninterrupt is received, f_(SRC) may be estimated:f _(SRC) =P/(t _(INTERRUPT[N+1]) −t _(INTERUPT[N]))

2. Compute nominal PWM counter values for N 181 & M 182 that willproduce a clock frequency as close as possible to the desired f_(MEDIA):(N+M)=f _(SRC) /f _(MEDIA)

3. Set the programmable counter to throw an interrupt every P=k*(N+M)cycles of f_(SRC) as shown in FIG. 6 . Each time this interrupt fires,the values of M 182, N 181 and P 180 are updated. For an audio system,preferably P 180 is chosen so that jitter introduced by clockadjustments is inaudible (e.g. a 1 Hz interrupt rate). Note that in eachclock update period, the clock can be corrected in multiples ofk/f_(SRC) seconds.

4. Initialise the residual synthesized clock error (to zero, say):E _(RESIDUAL)=0

5. Each time the interrupt fires:

a. Estimate the source clock rate f_(SRC) as in step 1.

b. Compute new (M+N)=f_(SRC)/f_(MEDIA).

c. Compute the error between the synthesized clock and the time of dayclock.t _(EXPECTED) =t _(INTERRUPT[N]) +k/f _(MEDIA)E=t _(EXPECTED) −t _(INTERRUPT[N+1])

d. Add computed error to residual synth clock errorE _(RESIDUAL) =E _(RESIDUAL) +E

e. Amortise residual clock error over the next k periods of the mediaclock(M+N)=(M+N)+quotient(E _(RESIDUAL)/(k/f _(SRC)))E _(RESIDUAL)=remainder(E _(RESIDUAL)/(k/f _(SRC)))

f. Program the PWM counter with the new values for N and M.

g. Update the value of P=k*(M+N).

h. Program the interrupt period with the new value of P.

(iii) Software-Only Systems

The software-only system approach for media clock synthesis isapplicable in systems that do not discipline a hardware media clock 123and 125. A software master clock 102 is a timer implemented in software.The timer is a common feature of operating systems. An application mayrequest to be signaled after T microseconds have elapsed, or may requestto be signaled at a particular (future) TOD instant. An example is ageneral purpose computer sending packetised audio data coming from afile over a network. The “media clock” in these systems is the timerused to schedule the transmission of the next media packet. It will beapparent that a software timer may be caused to track the synchronizedlocal TOD clock.

One approach for implementing such a system is:

-   -   1. T=1/f_(S)—the period of the sample clock        -   S_(PP)=number of samples per packet (e.g. 10)        -   Record a start time t_(START)=Read-TOD-current-value( )        -   Initialize a packet counter n_(PKTS)=0    -   2. Set timer to fire in S_(PP)*T seconds from now    -   3. [TIMER FIRES]        -   Send a packet with S_(PP) samples of audio data            n _(PKTS) =n _(PKTS)+1            t _(EXPECTED) =t _(START) +n _(PKTS) *T*S _(PP)            t _(NOW)=Read-TOD-current-value( )    -   4. Set timer to fire in S_(PP)*T−(t_(NOW)−t_(EXPECTED)) seconds        from now    -   5. Back to step 3 when the timer fires.

In many systems, software timers are not always guaranteed to arrivepromptly. If such a software timer arrives late, packets may have notbeen sent because the timer signal did not arrive. In this case thesystem “catches up” by sending the packets that were not sent. It willbe apparent that a system in which timers may fire later than onepacket-time from when they are scheduled to arrive will need to sendadditional packets at step 3 from time to time to catch up after latetimer arrivals.

Media Clock Offset Synchronisation

Media clocks offset synchronisation is achieved by:

1. counting ticks of the media clock since a nominated start time,

2. periodically computing a media clock absolute time(T=start-time+N*media_clock_period),

3. comparing the media clock absolute time with the master clock time toproduce a time difference, and

4. increasing or decreasing the media clock frequency to minimise thetime difference.

A wide variety of known phase locked loop techniques may be employed toincrease or decrease the media clock frequency and so implement thiscontrol loop. The process of counting media clock ticks since adesignated start time assigns an absolute time to each edge of the mediaclock and the control loop acts to ensure that the media clock edges arealigned with the master clock.

To illustrate, the counter 202 in FIG. 3 counts pulses of f_(LOCAL) (amultiple of f_(MEDIA)). This count in combination with a start time isused to produce a local TOD clock. Timestamps taken using the local TODclock are then compared to timestamps from the master clock in thenetwork time protocol messages 121 and the difference used to update theDDS tuning word (TW) 200 thus varying media clock frequency and offset.

Jitter Constraining QoS Scheme

Further to clock synchronisation, a QoS scheme can be used to controloverall jitter and delay in a network. Typical components of a medianetworking QoS scheme technique used for this invention include:

1. Classification of Packets According to Priority

This can be implemented by inspecting packet fields (e.g. 802.1Qpriority bits, IP Diffserv Code Point, UDP or TCP port numbers) set bythe source. Packets not labeled with a priority value are classified as“best effort” traffic.

2. Expedited Forwarding of High Priority Packets

In this approach, high priority packets are transmitted preferentially,minimizing the time they spend queued in switches of the network 106.Various mechanisms are commonly available including “Strict Priority”and “Weighted Fair Queuing” scheduling. A source may also implement apriority scheduler whereby it ensures that high priority packets aretransmitted preferentially.

3. Preventing Over-Subscription of the Network Using Admission Control

To ensure a given upper bound for jitter and to ensure that the networkis operated within total available capacity, bounds on maximum networkutilization of each traffic class are enforced. Clients may use asignaling protocol such as RSVP to discover whether the network willsupport additional network flows before transmitting packets into thenetwork. In simple network topologies where a network of switches isconnected via an over-provisioned backbone, the link between the clientand the next hop switch is the bottleneck. In this case the client mayperform a local admission check without needing to use hop by hop RSVPsignaling

Three approaches for QoS techniques in a network transmitting digitalaudio data are:

1. Three levels of priority—no NTP/1588 support in the switch.

Highest Priority: network time protocol traffic (NTP/1588)

Middle Priority: packets containing digital audio data

Lowest Priority: best effort traffic

Expedited forwarding is enabled using Weighted Fair Queuing or StrictPriority. Highest priority clock packets may still experience somejitter due to queuing behind low priority packets already in transit ina switch. While filtering of jittered packets can be used to improveperformance, this reduces the clock synchronization accuracy achievable.The network diameter (maximum number of hops) is typically restricted tostay within desired clock synchronization accuracy, end-to-end latencyand accuracy of sample alignment in the network. This may only berequired for certain critical paths of the network with laxer timingrequirements being acceptable on other paths. The bandwidth requirementfor an audio stream can be calculated knowing its bit depth and samplerate. The admission control mechanism is used to ensure that thebandwidth required by the sum of all streams traversing any particularnetwork link does not exceed the link capacity.

2. Two levels of priority—1588 border clocks in switches

If the network switch supports the IEEE 1588 network protocol, it willconsume and regenerate the network time protocol packets rather thanforwarding them. In this case, only two priority levels are needed:

Highest Priority: packets containing digital audio data

Lowest Priority: best effort traffic

Expedited forwarding is enabled using Strict Priority or Weighted FairQueuing. The bandwidth requirement for an audio stream can be calculatedknowing its bit depth and sample rate. The admission control mechanismis used to ensure that the bandwidth required by the sum of all streamstraversing any particular network link does not exceed the linkcapacity.

3. Over-provisioned, audio-only network.

For networks with few devices, few audio channels and constructed usinghigh speed links, such as a set of devices connected to a single gigabitEthernet switch, the network has more capacity than is required.Provided the network is only carrying audio and network time protocoltraffic, priority schemes, expedited forwarding and admission controlare not necessary. This is because the network is so lightly loaded thatit does not introduce significant jitter.

The above list of QoS schemes for transmitting digital audio data is notexhaustive, and may be extended to support video traffic or multipleclasses of audio traffic with different latency requirements by addingadditional priority levels.

Playout Time

Receivers 110 compute a playout time by adding a latency time interval,compensating for network jitter and timing errors, to the sourcetimestamp. The additional latency required can be easily measured ateach receiver 110 by examining the incoming timestamps from eachdistinct source to the local clock 124. Pre-recorded data can be sentearly, relying on the receiver to use the timestamp to compute thecorrect playout time. Sequences of event based data, such as MIDI orlighting control commands, can be supported.

An appropriate playout time can be represented as NOW+D. The value for Dcan by estimated by recording the worst case difference between the timeat which a packet was received, as measured by taking a timestamp usingthe local clock at the instant the packet arrives, and the timestampplaced in the packet by the sender 108.

Pre-recorded audio data, or event sequences can be sent with a timestampin the future, enabling them to be buffered by the receiver 11Q untilthe playout time arrives. For example, consider lighting control. Themessage TURN_ON can be sent to three different lights, one after theother, with the same timestamp, 1 second in the future from now. Eachlight will buffer the message until the playout time arrives, andalthough the messages were sent at different times, the lights will allturn on at the same time. In another example, a sequence of TURN_ONmessages can be sent to a light with a timestamp of NOW+1 s, andimmediately another message is sent with a TURN_OFF message to the samelight, but with a timestamp of NOW+10 s. The light processes eachmessage when the playout time arrives, turning the light on for 9seconds at one second after the messages were sent. In this example, thelighting control messages are sent “early,” with timestamps in thefuture.

Jitter Compensation Data Buffers 142 and 148

The playout time for a particular packet must take account of networktransmission delays, network time protocol synchronization errors, mediaclock synthesis errors, sender timer jitter and network jitter. Theplayout time of received packets must be delayed enough to allow forlate arrival of packets due to any of these causes. The length of thejitter compensation buffer 148 must be provisioned according to themaximum expected end-end delay between that sender 108 and receiver 110such that there is no buffer underrun. Packets arriving are stored inthe jitter compensation data buffer 148 until the playout time arrives.

For example, if packets may be delayed by at most D microsecondstraveling between the source and a receiver, the system can safelychoose a playout time of NOW+D microseconds. The packets that arereceived between NOW and NOW+D are stored in the buffer to ensure thatby the time NOW+D arrives, all delayed packets have been received.Playout of each packet occurs at D microseconds after the timestamp itcontains.

Each receiver 110 will independently determine an appropriate playouttime for audio from a particular sender 108. Under system control, tworeceivers 110 may choose to synchronize their playout times. For examplethe right and left channel in a home stereo would choose the sameplayout time for the right and left channel sound to be synchronized.Receivers 110 in different parts of the network also may choosediffering playout times. For example in a live concert, the delay fromthe performer to the speaker providing the musicians mix is critical,thus the network would be designed (i.e. by ensuring that high speedlinks are used and that the number of switch hops is minimised) suchthat the network delay in this path is minimal and the speaker uses anearlier playout time. On the other hand, the delay to the reinforcementspeakers in the hall is less critical and may involve more network hops.These receivers could use a later playout time than the musiciansspeaker. In addition, arrays of reinforcement speakers in a hall couldalso select different playout delays selected such that the phaserelationship between them makes for optimal listening.

Source timestamps can also be used to align and combine data fromdifferent sources even when the delays through their respective networkpaths to the receiver are different, for instance in audio mixing. Audioplayout may also be synchronized with non-streaming data such as MIDI byaligning the timestamps in the MIDI data with the timestamps in theaudio data. This same approach applies for any timestamped media, video,lighting control. A given receiver 110 may pick different playout timesfor each audio channel as described above. If the system is minimisingthe latency through the network 100 for each audio channelindependently, then the delay (D) can be estimated for each channel andthe playout time NOW+D can be calculated for each channel separately.The above functionality is not possible if the audio packets aretimestamped with a playout time, rather than the generated time.

An implementation on a general purpose computer system requires a jittercompensation latency of the order of 5-10 ms, primarily due to timerjitter in the sending system and NTP synchronization errors. The minimumjitter compensation latency for a tightly synchronized system (e.g. ahardware implementation using IEEE 1588) is one packet interarrivaltime—just as the current packet is consumed, the next packet arrives.

Since media packets are generated at regular intervals, if all sourcesconnected to a switch generate a packet at the same time, they would bepassed through the switch as a bunch. If this bunch of packets thenencounters a similar bunch of packets at the next switch hop, then thelatency through a switch would increase at each hop. To avoid thiseffect sender 108 randomises start times and thereafter generate packetsat a regular rate with respect to this random start time at the governedrate. This helps to reduce bottlenecks in the network 106.

More than one sender 108 is able to send packets over the network 106 atany given time—in keeping with the usual mode of operation of packetswitched networks like Ethernet. The maximum network latency encounteredby audio packets between any sender 108 and a receiver 110 may bybounded by using well understood Quality of Service (QoS) techniques.These include using admission control to limit the number of audiostreams on each link as well as deploying scheduling techniques inswitches to prioritise audio packets over other non-real time data asdiscussed above. Therefore a system with tight clock synchronizationwith network QoS should be able to achieve a smaller end-to-end latencythan a system with looser clock synchronization and no QoS. Animplementation on a general purpose computer system requires a jittercompensation latency of the order of 5-10 ms which is primarily due totimer jitter in the sending system and NTP synchronization errors.

The architecture of the invention also supports non-streamed data thatrequires time synchronization. Examples are MIDI and lighting control,as discussed next.

EXAMPLE

FIG. 7 is a block diagram of a typical application of the invention. Themaster clock device 102 produces a system time signal 104 which isrelayed to the network devices 108, 110 and 111. These network devicesmay be connected to a variety of media devices 112 and 114 as shownhere, or alternatively they may be incorporated into the media device.Some of the media devices may be digital sources 112 or analog sourcesmicrophones 170 and a guitar 172. Others may be audio playout devices114 such as speakers 174 and the redundant speaker 176. Still others maybe media processing devices such as a MIDI sound console 178, a mixer160 or a lighting control module 179. A computer 162 may be used torecord and send audio, and also to send MIDI and lighting commandsignals. Another computer 164 may be used as a configuration console forthe network 100.

The invention supports a wide range of application domains from low costdevices with lax timing requirements, such as audio distribution in anairport, to professional audio production systems requiringmulti-channel sample accurate timing. Further suitable applications ofthe invention include public address system, live music systems,recording studios including professional and home recording studios, andhome theatre systems.

Advantages of the Invention

Reduced cabling costs—Using this invention, each cable of the network isable to carry multiple channels, which leads to reduced cablingcomplexity and cost. Further, the cables may contain a mix of differentspeed links, such as low speed device links and a highspeed backbone. Asimple interface on a computer may be provided for equipment needingmany channels, providing reduced setup time, including plug and playconnection.

The “audio snake” is a common component in audio systems. It is a bundleof audio cables wrapped in a sheath and is terminated with a sprout ofconnectors or a patch bay. There is a one-to-one correspondence betweensockets on the patch bay and connectors in the sprout. An audio snake isan expensive cable due to the high cost of the connectors and the labourinvolved terminating the many wires. Using Ethernet vastly reduces thecabling cost and reduces the number of connectors.

Speaker cabling carries high-power audio signals. Speakers are fairlylow impedance devices (typically 4 or 8 Ohm) and high cable impedancescause power losses and changes the frequency response of the speaker.Every metre of cable in a long run adds to the cable impedance. Lowimpedance speaker cable is thick and expensive. The closer the poweramplifier is to the speaker the lower the losses. Placing the poweramplifier inside the speaker is increasingly common, but requires powerand the audio signal to be routed to every speaker. Digital audionetworking as used by the invention is a scalable way of routing audiosources to powered speakers.

Easier maintenance—Using the invention, moving a piece of equipment fromone network port to another port does not require audio routingconfiguration to be changed. This is in contrast to audio networkingsolutions where audio routing is configured hop-by-hop at each “switch.”Changing the location of a device in such systems involves eitherre-cabling to make it appear that the device is still in the oldlocation, or reconfiguring the “switches” to send the audio to the newlocation, rather than the old location. The “switches” in such systemsdo not conceptually route packets (e.g. as an Ethernet switch would do),rather they route audio channels.

Producing an audio track is not a set and forget affair. A soundengineer at a mixing console adjusts the levels of each instrument,fades between different tracks and adjusts effects during each track toget the best combined sound from the recorded sources. These adjustmentscan be considered part of the final performance. Recording and replayingthe actions of the audio engineer avoids tedious repetition and reduceserrors. An audio network facilitates the transport of automated orrecorded mixing commands to equipment attached to the network (e.g.equalisers, effects processors).

Scalability—At least one embodiment of the invention provides amulticast network architecture that can be scaled to thousands ofreceivers, supporting large venues like sporting stadia.

Flexibility—The invention may be implemented in a variety of differentsituations as described above. This includes, high quality hardwareimplementations that require low latency and sample accurate timing.Sample accurate timing is the capability of two (or more) devicesattached to a network to play out an audio sample at the same time—towithin a single sampling period of the digital audio. For example, leftand right speakers must play out audio in synchrony otherwise poor soundreproduction will occur. Sample accurate timing ensures that digitalaudio data from the left and right channels having the same timestampwill be played out the left and right speakers at the same time, towithin the time period for a single sample. Samples with the sametimestamp will be “aligned in time” at the moment they are played out onall devices connected to the network. At the highest sample ratessupported by audio systems today (192 kHz), this translates to a timesynchronisation accuracy of about 1 microsecond. On the other handcost-conscious software implementations with relaxed timing can stillsend and receive audio data into the network.

Allowing different audio mixes—The invention has applications insituations where there are different audio mixes coming from differentspeakers, such as “Audience Mix” and “Musicians Mix.” For example,musicians performing live (e.g. a rock band) commonly use amplificationto boost the level of sound for the audience. The “Audience Mix” is anaudio signal containing all of the instruments and vocals which isplayed out the speakers to the audience. The “Audience Mix” is sometimescalled the “Front of House” mix. The musicians, however, also need tohear themselves clearly to play well. The “Musicians Mix” is the audiosignal amplified and played out the speakers pointing at the musicianson stage. Often, each musician has an individual speaker and mix inwhich their instrument is louder than the others. Using the invention atenfold increase in bandwidth leads to a tenfold decrease in latency,and a tenfold increase in the number of channels.

By using networks such as Ethernet networks, the invention allows directinterfacing with computer equipment. Standard computers can participatein the network timing protocol and remain in synchronization with otherdevices in the network. Use of UDP/IP for audio data encapsulationallows interfacing with computer systems. Using the computer interfacemoving a piece of equipment is also simplified.

In traditional systems, each digital audio link type has a differentdata format, different cabling and different plugs. The result is thataudio equipment usually has a variety of different connectors and plugsto transport the same basic PCM encoded audio data. Each component of anaudio system must have an intersecting set of connectors with anothercomponent in the audio system. Each component in the system usuallyneeds to be configured with basic information like sample rate and bitdepth. Digital audio networking as used by the invention provides a wayof transporting audio streams of different types over a common network.Further, the same network can be used for transporting network timeinformation.

Support for MIDI Data—The Musical Instrument Digital Interface isreferred to as MIDI. Pressing and releasing keys on an electronic pianogenerates MIDI messages describing the pitch, duration and velocity ofeach note played. MIDI is a popular interconnection technology now usedin areas other than instrument control. Today it is used to controlaudio effects units (reverberation, equalisation, etc), lightingequipment, and for configuration parameters for musical equipment. Thisinvention supports the transmission of event data, like MIDI, over asingle audio network.

Accurate Timing for MIDI—The MIDI standard specifies a 31.25kbaud/second serial cable supports 16 channels. Low bus bandwidth hastwo effects: first it limits the number of control events that can besent, and second, the timing of simultaneous events is impaired as thebus load increases. These two issues are routinely avoided by usingmultiple MIDI busses.

Precise timing of MIDI events output from each port is achieved by clocksynchronisation between the PC and the MIDI box, and by embedding timinginformation in the MIDI events traveling over, e.g. a USB bus. A digitalaudio network can replace the USB bus, and can further directlytransport MIDI events to the devices that consume them with accuratetiming.

Timing Errors—Timing errors can result in phase errors in audio output.The human ear is highly sensitive to phase errors and can detect changesin sounds produced by timing errors of less than a millisecond. Theinvention supports precise time timing for MIDI and audio data allowingtight synchronisation to be achieved.

Peer-to-peer capability—Current audio systems are centralised. Allsource audio data is usually routed to a mixing desk, processed and thenprovided to speakers. A peer-to-peer audio network architecture removesthe requirement for a centralised mixing desk—audio sinks can locate andconsume audio data directly.

A peer-to-peer digital audio network architecture also makes recordingof digital audio data simpler. A recorder can locate and sink allsources of audio data in the network. This is enabled for a standalonerecording device or PC-based audio production software.

Data networking technologies focus on error free reliable communication.As an example, Ethernet boasts high bandwidth, low bit error rates andlow cost cabling. In addition, robustness against component failure isimportant in critical communication systems. A wide variety of protocoltechniques for failing over between redundant links and services areavailable.

Although the invention has been described with reference to particularexamples and applications it should be appreciated that it may be putinto effect in many other forms and for many other applications.

It will be appreciated by persons skilled in the art that numerousvariations and/or modifications may be made to the invention as shown inthe specific embodiments without departing from the spirit or scope ofthe invention as broadly described. The present embodiments are,therefore, to be considered in all respects as illustrative and notrestrictive.

What is claimed is:
 1. A data network suitable for transporting videopackets, the network comprising: a master clock device to generate asystem time signal for the network; a plurality of network devicesinterconnected such that at least one network device can send videopackets to other network devices, and to receive video packets fromother network devices; wherein at least one network device is configuredto receive video signals from a media device and to transmit videosignals to a media device; and wherein, the network devices are coupledto the master clock device and use the system time signal and a networktime protocol to generate a local clock signal synchronized to thesystem time signal for both rate and offset, the local clock signalgoverning both the rate and offset of the received or transmitted videosignals.
 2. A data network according to claim 1 wherein, in use, thenetwork devices that receive media signals from a media device use itslocal clock signal to packetize the video signals, and to timestamp thevideo packets with system time before transmitting them to anothernetwork device.
 3. A data network according to claim 2 wherein thetimestamp of a video packet includes the earliest system time at whichthe video signal contained within the video packet was generated.
 4. Adata network according to claim 1 wherein the network devices thatreceive video packets use the packet timestamps to reorder video packetsas required, and in addition to perform at least one of (1) to align andcombine video signals received from different network devices, and (2)to determine the appropriate playout time for video signals contained inthe video packets.
 5. A data network according to claim 4 wherein theappropriate playout time for video signals contained in the videopackets is different for two network devices that receive the videopackets.
 6. A data network according to claim 1 wherein the video mediapackets contain both audio and video media signals.
 7. A data networkaccording to claim 1 wherein the network comprises an Ethernet network.8. A data network according to claim 1 wherein a media clock signal issynthesized from at least the local clock signal, such that the mediaclock signal directly governs the rate and offset at which video signalsare received or transmitted.
 9. A data network according to claim 8wherein the rate of the media clock signal of a network device isdifferent from the rate of a media clock signal of another networkdevice.
 10. A data network according to claim 8 wherein the media clocksignal is used for conversion between analogue and digital mediasignals.
 11. A data network according to claim 8 wherein the media clocksignal is synthesized by one of pulse width modulated clock synthesisinvolving a pulse width modulated counter, direct digital synthesis, adigitally controlled oscillator, a voltage controlled oscillatorcontrolled by a digital to analogue converter, and a software timercaused to track a synchronized local clock signal.
 12. A data networkaccording to claim 1 wherein the rate of the local clock signal of anetwork device is different from the rate of a local clock signal ofanother network device.
 13. A data network according to claim 1 whereinat least one media device is connected to a network device for carryingout at least one of (1) to generate and deliver video signals, (2) toreceive, to process and to deliver video signals, and (3) to receive andplayout video signals.
 14. A data network according to claim 1 wherein aquality of service scheme of the data network operates to prioritize thetransmission of packets that contain network time information.
 15. Adata network device for connection to other network devices to transportvideo packets between them, the network device comprising: a media portfor receiving video signals from a media device and transmitting videosignals to a media device; a clock port coupled to receive a system timesignal; and a local clock generator coupled to use the system timesignal and a network time protocol to generate a local clock signalsynchronized to the system time signal for both rate and offset, thelocal clock signal governing both the rate and offset of the received ortransmitted video signals.